CUCM with Microsoft Unified Messaging Call Transfer issue

[Kapil Atrish]

Hi,

I've a customer using CUCM with Microsoft Unified Messaging. When IP Phone user sets Call Forward all to voice-mail, the transferred calls never reach his voice-mail box. The caller gets dead air. Direct calls to the extension does get forwarded to voice-mail correctly.

Consider a scenario when IP Phone user 2001 sets CFWD ALL to voice-mail. Anyone ringing 2001 directly reaches his voice-mail box but when operator transfers anyone to 2001, the caller gets dead-air instead of voice-mail prompt.

Did some research, couldn't find any information on this behavior. Tried looking for Service Parameter but couldn't see any relevant settings for Transferred calls.

Any pointers on this issue will be greatly appreciated.

thanks,
inner_silence

[Amer rajai Sha'er]

Hello,

Does the CFWDALL on the extension configuration has a CSS that can dial the through the trunk??

Amer

[kking2008]

A few things to check/consider

1. MS UM will ONLY communicate G.711ulaw, it will not use G.729r8.  So be aware of your Region settings between source and destination.

2. Make sure the CSS's for redirecting numbers is set as well.  MS UM will need the redirected number in order to put the caller in the right mailbox.

[Kapil Atrish]

Hi King,

I tried, don't see Re-routing CSS settings, where do you set it exactly?.I am forwarding the calls to voice-mail via "Forward to VM" check-box so its going to take the CSS on Line/Phone to route the Pilot. That's  how the IP Phone is able to dial the voice-mail pilot, when someone calls in, the caller is able to get directed to voice-mail. Its just that when Operator/any other IP Phone transfers the call to the phone in quesiton, the caller isn't able to reach the voice-mail box. Caller gets dead-air. Internal transferred calls are also seeing the same behavior.

Thanks in advance,

inner_silence

[kking2008]

I was referring to the setting in the SIP trunk to ExchangeUM server.  Near the bottom of the page you should find

Rerouting Calling Search Space

This CSS will need to be set appropriately.

Rerouting Calling Search Space

Calling search spaces determine the partitions that calling devices can search when they attempt to complete a call. The rerouting calling search space gets used to determine where a SIP user (A) can refer another user (B) to a third party (C). After the refer is completed, B and C connect. In this case, the rerouting calling search space that is used is that of the initial SIP user (A).

good luck

[Kapil Atrish]

Hi,

I checked, the Rerouting CSS has access to VM Pilot/SIP Trunk. Have you tested this scenario with Microsoft UM, put call-forward all to VM, and then transfer a call to that extension and see if you get correct VM greeting or dead air like me. Let me know if there is anything else I should pay attention to.

Thanks in advance,

inner_silence

[Kapil Atrish]

I was able to figure it out. MTP wasn't enabled on SIP Trunk. Enabling it resolved the issue. Thanks for your efforts in helping me out.